Hi, yes, Asterisk will send media to RTPproxy IP and not to the UAC. Looking at the first trace you send, I see that rtpporxy was not set for the 200 OK reply (only for INVITE request) - because of this, the media is broken.
Regards, Bogdan oso che bol wrote: > Dear Bogdan, > > My current Problem is: My Asterisk VOICEMAIL app, /which opensips > foward to/, return voice announcement to UA OF OPENSIPS, and expected > results will be: we could here voice of that. But, actually result is > that we could not hear anything. > > I also attach trace of Asterisk communication with Opensips when > INVITE come out opensips to Asterisk. One notice that Invite come out > Opensips also involve RTPPROXY IP :(, maybe that is glue of my > problem? And we have 2 ACK from opensips to Asterisk at the last of trace. > > Thanks and Regards, > -LN > > ==========TRACE of ASTERISK============= > > filter: (ip) and ( port 5060 ) > > U OPENSIPS_IP:5060 -> ASTERISK_IP:5060 > INVITE sip:3...@asterisk_ip:5060 SIP/2.0. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480. > Max-Forwards: 69. > Contact: <sip:[email protected]:25480 > <http://sip:[email protected]:25480>>. > To: "3000"<sip:3...@opensips_ip>. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 INVITE. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO. > Content-Type: application/sdp. > User-Agent: X-Lite release 1100l stamp 47546. > Content-Length: 443. > P-hint: HTK: Fix nated contact. > P-hint: HTK: INVITE go to on_reply_route[1]. > P-hint: INVITE||ACK + FORCE_RTP_PROXY. > P-hint: HTK - 408 and route to VM ... . > . > v=0. > o=- 8 2 IN IP4 192.168.1.150. > s=CounterPath X-Lite 3.0. > c=IN IP4 *RTPPROXY_IP.* > t=0 0. > m=audio 42762 RTP/AVP 107 119 100 106 0 105 98 8 3 101. > a=alt:1 1 : pt1QlLaH Tih+tcui 192.168.1.150 50854. > a=fmtp:101 0-15. > a=rtpmap:107 BV32/16000. > a=rtpmap:119 BV32-FEC/16000. > a=rtpmap:100 SPEEX/16000. > a=rtpmap:106 SPEEX-FEC/16000. > a=rtpmap:105 SPEEX-FEC/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > a=nortpproxy:yes. > > > U ASTERISK_IP:5060 -> OPENSIPS_IP:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > To: "3000"<sip:3...@opensips_ip>. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:3...@asterisk_ip>. > Content-Length: 0. > . > > > U ASTERISK_IP:5060 -> OPENSIPS_IP:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP > OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > To: "3000"<sip:3...@opensips_ip>;tag=as39e194d1. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:3...@asterisk_ip>. > Content-Length: 0. > . > > > U ASTERISK_IP:5060 -> OPENSIPS_IP:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > To: "3000"<sip:3...@opensips_ip>;tag=as39e194d1. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:3...@asterisk_ip>. > Content-Type: application/sdp. > Content-Length: 289. > . > v=0. > o=root 9042 9042 IN IP4 ASTERISK_IP. > s=session. > c=IN IP4 ASTERISK_IP. > t=0 0. > m=audio 12610 RTP/AVP 3 0 8 101. > a=rtpmap:3 GSM/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U ASTERISK_IP:5060 -> OPENSIPS_IP:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > OPENSIPS_IP;branch=z9hG4bK8303.a0576353.1;received=OPENSIPS_IP. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-b676bc4d1975ea69-1---d8754z-;rport=25480. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > To: "3000"<sip:3...@opensips_ip>;tag=as39e194d1. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:3...@asterisk_ip>. > Content-Type: application/sdp. > Content-Length: 289. > . > v=0. > o=root 9042 9042 IN IP4 ASTERISK_IP. > s=session. > c=IN IP4 ASTERISK_IP. > t=0 0. > m=audio 12610 RTP/AVP 3 0 8 101. > a=rtpmap:3 GSM/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U OPENSIPS_IP:5060 -> ASTERISK_IP:5060 > ACK sip:3...@asterisk_ip SIP/2.0. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.3. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-ca5726034435a134-1---d8754z-;rport=25480. > Max-Forwards: 69. > Contact: <sip:[email protected]:25480 > <http://sip:[email protected]:25480>>. > To: "3000"<sip:3...@opensips_ip>;tag=as39e194d1. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 ACK. > User-Agent: X-Lite release 1100l stamp 47546. > Content-Length: 0. > P-hint: HTK: rr-enforced. > P-hint: Route[1] Processing. > . > > > U OPENSIPS_IP:5060 -> ASTERISK_IP:5060 > ACK sip:3...@asterisk_ip SIP/2.0. > Record-Route: <sip:OPENSIPS_IP;lr=on;ftag=43241208>. > Via: SIP/2.0/UDP OPENSIPS_IP;branch=z9hG4bK8303.a0576353.3. > Via: SIP/2.0/UDP > 192.168.1.150:29280;received=118.69.139.66;branch=z9hG4bK-d8754z-ca5726034435a134-1---d8754z-;rport=25480. > Max-Forwards: 69. > Contact: <sip:[email protected]:25480 > <http://sip:[email protected]:25480>>. > To: "3000"<sip:3...@opensips_ip>;tag=as39e194d1. > From: "6000"<sip:6...@opensips_ip>;tag=43241208. > Call-ID: ZWZhM2I2YmRiMzkzNThkNDhlMzdjMjk5MzA0Y2I1YTk.. > CSeq: 1 ACK. > User-Agent: X-Lite release 1100l stamp 47546. > Content-Length: 0. > P-hint: HTK: rr-enforced. > P-hint: Route[1] Processing. > . > > > > On Thu, Apr 9, 2009 at 4:25 PM, Bogdan-Andrei Iancu > <[email protected] <mailto:[email protected]>> wrote: > > Hi, > > The script and signalling are ok and looks good. Maybe you can > detail a bit what you do not like and what you want to change. > > Regards, > Bogdan > > oso che bol wrote: > > Dear All, > > Platform: > - Opensip 1.4.5 > - RTPPROXY: 1.2.0 > - MySQL: 5.0 > - Asterisk 1.4.24 > > Asterisk acts as a VoiceMail system when Users on Opensips Do > Not answer or Busy, so call will route to Asterisk to leave > Voicemail. > Voicemail system works properly. > > The problem is: "When call route out of Opensips Server to > Asterisk, Request URI use IP of Opensips to request Asterisk". > > So, Asterisk will return to wrong IP of UA Client, and of > course, no voice from Asterisk to UA Client. > > Expected Results:1. "I want to use 1...@ip_of_uaclient to > request Asterisk for leave voicemail?" or "If it uses > 1...@ip_of_opensips to request Asterisk for Voicemail, UA > Client should hear Voice". > > Bellow is my Configuration Files and Log File of OpenSIPs. > > Thanks and Regards, > -LN > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
